To provide the best quality audio, a general principle is to apply as few operations as possible to signals in the signal path from an audio signal source to the audio signal destination, i.e. the eventual audible signal, whether these signals and operations are digital and/or analog. Each stage of processing may introduce audio artifacts such as added truncation or thermal noise to the signal, or added non-linear signal-related components, i.e. distortion, to the signal, or may give a frequency response that is not flat or does not correspond exactly to some desired optimal frequency response. Also for time-sampled signals, the sample rate should generally be as high as is practicable.
In many instances an audio reproduction apparatus may have at least one mode of operation in which audio data received, e.g. the ‘raw’ audio data from a storage medium, may be subject to minimal processing, for example a single-bit pulse-density-modulated (PDM) bit stream may be received and passed through a simple one-bit-input digital-to-analog convertor (DAC) and through a buffer amplifier to drive an audio output transducer such as a loudspeaker. The amplifier may have controlled gain to provide a volume control, i.e. the gain is set by a received volume control signal.
However in other cases, it may be desired to perform some operation on the input signal, for example mixing in other signals, or applying some deliberate frequency response weighting or applying a user-controlled or automatic gain control in the digital domain. For instance, an electronic device such as a smart phone may allow playback of stored audio data, such as music. If some event occurs during music playback that would be indicated by an audible alert, e.g. a tone to indicate receipt of a message or a status alert such as low battery, such a tone may be mixed into the music signal being output.
FIG. 1 shows an example where one component 100 of an audio system, which may be circuit, e.g. a single integrated circuit or a plurality of integrated circuits, or a functional unit or module, may in a first mode of operation provide via a first signal path an audio bit stream DSDin, say a one-bit oversampled audio bit stream, to a second component 101 comprising a buffer amplifier 102 with a controllable gain A2. In an alternate mode of operation, implemented via a second signal path, the audio bit stream DSDin may be first decimated to provide a multi-bit PCM word at a lower sample rate, which is then subject to a digital gain Apcm and possibly mixed with or substituted by some other signal, for example a ring tone for a wireless telephone. The resulting signal PCMout is then transmitted to the second component 101, and passed through the buffer amplifier 102, which in this mode may be set at a fixed gain. Buffer amplifier 102 may be a digital-input buffer amplifier, for example a digital PWM modulator, or may incorporate a digital-to-analog converter and an analog buffer amplifier. The signal PCMout may be re-converted into a one-bit (or multi-bit) oversampled audio bit stream before being input to the second component 101.
The first component 100 may for example be an applications processor (AP) in a wireless telephone handset. The second component 101 may be a codec integrated circuit inside the handset, or perhaps at the remote end of an attached accessory such as a headset, in either case perhaps switchably connected to the applications processor via a switched connection at a socket or plug or similar.
There can, in practice, be several problems with this arrangement.
A volume control signal controlling any gain to be applied by the buffer amplifier 102 in the first mode of operation will generally be generated in the applications processor. However the transmission link from the applications processor to the codec may not readily allow transmission of this control information. In some instances the volume control signal may be generated at a high level of the operating system of the application processor, for example within some Android™ subsystem, and there may be unacceptably large delays, perhaps of the order of 300 ms, in transferring the volume control signal to the second component. These delays may be problematic, especially say if the control signals comprise a long ramped sequence of ramped gain by single-step increments.
Similarly the second component 101 also requires some signal to control whether to accept DSDin or PCMout as its input signal for processing, and which of the respective appropriate gains to apply.
Another issue is illustrated with respect to FIGS. 2a and 2b. Assuming an appropriate sample rate and noise shaping of the received oversampled audio bit stream DSDin, the noise level apparent in the audible output may be dominated by the thermal noise of the driver amplifier, i.e. buffer amplifier 102.
In FIG. 2a, which illustrates a model of noise in the first mode of operation using the first signal path, this noise source is represented by noise signal Vn added to DSDin (DSDin being regarded as an equivalent analog signal) at the input to the gain-controlled amplifier 102, which amplifier may then be assumed noiseless. Regardless of the volume setting, a (near) full-scale audio signal at Vin will still appear (near) full-scale when added to Vn. The peak-signal-to-noise ratio at the input to the noiseless amplifier is thus the full-scale signal divided by the noise Vn. The combined signal is then regarded as passing through the noiseless amplifier stage, which will scale signal and noise together and thus not affect the overall signal-to-noise.
In FIG. 2b, which illustrates a model of noise in the second mode of operation using the second signal path, the received DSDin signal is down-sampled and passed through a digital multiplier, before transmission to the second component and being added (regarded as an equivalent analog signal) to the equivalent noise source Vn. In contrast to the potentially full-scale signal added to the noise in FIG. 2a, the maximum level of the signal PCMout will be some fraction A of the full-scale of the PCM digital signal PCMout. Thus the signal-to-noise ratio will be a factor 1/A worse in this second mode of operation than the case for the first mode of operation as illustrated in FIG. 2a. The driver amplifier 102 may be required to drive say 1 Vrms into a line load with the amplifier at “unity” gain, whereas a typical peak signal into a headphone speaker may be 100 mV rms or more likely 10 mVrms, so the gain A may typically be 1/10 or even 1/100 so this degradation in signal-to-noise may be objectionable.